1. Field of the Invention
The present invention relates to an IP telephony apparatus providing simultaneous communication for multiple IP phones and a method for the same, especially to an IP telephony apparatus providing simultaneous SIP communication for multiple IP phones by using only one SIP number and a method for the same.
2. Description of Prior Art
The progress of Internet technology provides innovative ideas and new services. For example, the VoIP (Voice over Internet Protocol) technique provides telephony communication through IP network and the expensive long-distance call fee can be saved. More particularly, the VoIP technique provides PC-to-PC telephony communication through IP network, PC-to-Phone telephony communication through PBX (private branch box), Phone-to-Phone telephony communication through ISP gateway and Device-to-Device telephony communication through IP Phones.
The VoIP technique provides suitable telephony signal and voice transmission for conveying phone call through IP network. The VoIP technique sends telephony signal with specific protocol to represent user status and to establish communication for user. Once the communication is established for user, the voice is compressed and digitalized to transmit in the form of digital signal.
The conventional telephony signals such as dialing signals, ringing signal, and busy signal are converted into data packets according to VoIP protocol and the data packets a resent to remote user through IP network. The data packets are then converted to analog telephony signal by remote IAD or ATA for the operation of remote telephone set.
After the connection is established, the analog voice is sent to a local router through a telephone set, a fax or a PBX. The analog voice is compressed and digitalized into data packets and the data packets are sent to remote router through IP network. The data packets are converted to analog voice signal by the remote router and then sent to remote user through a telephone set, a fax or a PBX. The user can make cheaper long-distance phone call using VoIP technique through omnipresent IP network instead of the conventional PSTN system.
The present VoIP technique is regulated by ITU (International Telecommunication Union) and the earlier protocol such as H323/H248 are defined for LAN rather than the open environment of Internet. Therefore the H323/H248 protocol has limited application and complicated conversion for PSTN system. A new protocol, namely SIP (Session Initiation Protocol) protocol, is defined by IETF (Internet Engineering Task Force) to fully exploit the Internet service and provide better integration of Internet and PSTN system.
The SIP protocol belongs to the application layer protocol in the seven-layer architecture of the OSI (Open System Interface) and is resemblant to the Client-Server structure in HTTP protocol. Therefore, the SIP protocol can utilize existing HTTP packet structure for sending command and data and can be adapted for data transmission in WAN.
In present VoIP telephone system, a UA ((User Agent) such as a VoIP gateway is need to install at user side and at least one call server should be installed at VoIP agent. Moreover, the VoIP user needs to register an SIP VoIP number to the VoIP agent. Therefore, other VoIP user can call him through the SIP VoIP number.
However, the nowadays VoIP gateway generally uses VoIP H323/H248 protocol and a VoIP telephone set such as a USB telephone set is required. When the VoIP gateway is connected to multiple VoIP telephone sets, each of the VoIP telephone set needs a unique SIP VoIP number to prevent blocked call. Therefore, the conventional VoIP gateway needs at least two SIP VoIP numbers for two VoIP telephone sets to prevent blocked call.